[Asterisk-Users] Asterisk > Asterisk > PSTN

[Daniel Starks]

Hello all,

I have a sip client that is register on one asterisk
server, that asterisk server is routing the sip call
to another asterisk server where it hops off to a pstn
line via a X100P card. The call goes out but there is
no audio on either side. I have checked the codecs on
both servers to insure that that is not the issue, but
I have been unable to find the problem. If someone
might know the direction to look in i would appreciate
a point in that direction.

Thanks in advance for all assistance.

Daniel Starks
dlstarks at yahoo.com


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